The latency is the time it takes for a sample captured at timestamp 0 to reach the sink. This time is measured against the pipeline's clock. For pipelines where the only elements that synchronize against the clock are the sinks, the latency is always 0, since no other element is delaying the buffer.

For pipelines with live sources, a latency is introduced, mostly because of the way a live source works. Consider an audio source, it will start capturing the first sample at time 0. If the source pushes buffers with 44100 samples at a time at 44100Hz, it will have collected the buffer at second 1. Since the timestamp of the buffer is 0 and the time of the clock is now >= 1 second, the sink will drop this buffer because it is too late. Without any latency compensation in the sink, all buffers will be dropped.

The situation becomes more complex in the presence of:

  • 2 live sources connected to 2 live sinks with different latencies

    • audio/video capture with synchronized live preview.
    • added latencies due to effects (delays, resamplers…)
  • 1 live source connected to 2 live sinks

    • firewire DV
    • RTP, with added latencies because of jitter buffers.
  • mixed live source and non-live source scenarios.

    • synchronized audio capture with non-live playback. (overdubs,..)
  • clock slaving in the sinks due to the live sources providing their own clocks.

To perform the needed latency corrections in the above scenarios, we must develop an algorithm to calculate a global latency for the pipeline. This algorithm must be extensible, so that it can optimize the latency at runtime. It must also be possible to disable or tune the algorithm based on specific application needs (required minimal latency).

Pipelines without latency compensation

We show some examples to demonstrate the problem of latency in typical capture pipelines.

Example 1

An audio capture/playback pipeline.

  • asrc: audio source, provides a clock
  • asink audio sink, provides a clock
| pipeline                 |
| +------+      +-------+  |
| | asrc |      | asink |  |
| |     src -> sink     |  |
| +------+      +-------+  |

    • asink: NULL→READY: probes device, returns SUCCESS
    • asrc: NULL→READY: probes device, returns SUCCESS

    • asink: READY:→PAUSED open device, returns ASYNC
    • asrc: READY→PAUSED: open device, returns NO_PREROLL
  • Since the source is a live source, it will only produce data in the PLAYING state. To note this fact, it returns NO_PREROLL from the state change function.

  • This sink returns ASYNC because it can only complete the state change to PAUSED when it receives the first buffer.

At this point the pipeline is not processing data and the clock is not running. Unless a new action is performed on the pipeline, this situation will never change.

  • PAUSED→PLAYING: asrc clock selected because it is the most upstream clock provider. asink can only provide a clock when it received the first buffer and configured the device with the samplerate in the caps.

  • sink: PAUSED:→PLAYING, sets pending state to PLAYING, returns ASYNC because it is not prerolled. The sink will commit state to PLAYING when it prerolls.

  • src: PAUSED→PLAYING: starts pushing buffers.

  • since the sink is still performing a state change from READY→PAUSED, it remains ASYNC. The pending state will be set to PLAYING.

  • The clock starts running as soon as all the elements have been set to PLAYING.

  • the source is a live source with a latency. Since it is synchronized with the clock, it will produce a buffer with timestamp 0 and duration D after time D, ie. it will only be able to produce the last sample of the buffer (with timestamp D) at time D. This latency depends on the size of the buffer.

  • the sink will receive the buffer with timestamp 0 at time >= D. At this point the buffer is too late already and might be dropped. This state of constantly dropping data will not change unless a constant latency correction is added to the incoming buffer timestamps.

The problem is due to the fact that the sink is set to (pending) PLAYING without being prerolled, which only happens in live pipelines.

Example 2

An audio/video capture/playback pipeline. We capture both audio and video and have them played back synchronized again.

  • asrc: audio source, provides a clock
  • asink audio sink, provides a clock
  • vsrc: video source
  • vsink video sink
| pipeline                 |
| .------.      .-------.  |
| | asrc |      | asink |  |
| |     src -> sink     |  |
| '------'      '-------'  |
| .------.      .-------.  |
| | vsrc |      | vsink |  |
| |     src -> sink     |  |
| '------'      '-------'  |

The state changes happen in the same way as example 1. Both sinks end up with pending state of PLAYING and a return value of ASYNC until they receive the first buffer.

For audio and video to be played in sync, both sinks must compensate for the latency of its source but must also use exactly the same latency correction.

Suppose asrc has a latency of 20ms and vsrc a latency of 33ms, the total latency in the pipeline has to be at least 33ms. This also means that the pipeline must have at least a 33 - 20 = 13ms buffering on the audio stream or else the audio src will underrun while the audiosink waits for the previous sample to play.

Example 3

An example of the combination of a non-live (file) and a live source (vsrc) connected to live sinks (vsink, sink).

| pipeline                 |
| .------.      .-------.  |
| | file |      | sink  |  |
| |     src -> sink     |  |
| '------'      '-------'  |
| .------.      .-------.  |
| | vsrc |      | vsink |  |
| |     src -> sink     |  |
| '------'      '-------'  |

The state changes happen in the same way as example 1. Except sink will be able to preroll (commit its state to PAUSED).

In this case sink will have no latency but vsink will. The total latency should be that of vsink.

Note that because of the presence of a live source (vsrc), the pipeline can be set to playing before the sink is able to preroll. Without compensation for the live source, this might lead to synchronisation problems because the latency should be configured in the element before it can go to PLAYING.

Example 4

An example of the combination of a non-live and a live source. The non-live source is connected to a live sink and the live source to a non-live sink.

| pipeline                 |
| .------.      .-------.  |
| | file |      | sink  |  |
| |     src -> sink     |  |
| '------'      '-------'  |
| .------.      .-------.  |
| | vsrc |      | files |  |
| |     src -> sink     |  |
| '------'      '-------'  |

The state changes happen in the same way as example 3. Sink will be able to preroll (commit its state to PAUSED). files will not be able to preroll.

sink will have no latency since it is not connected to a live source. files does not do synchronisation so it does not care about latency.

The total latency in the pipeline is 0. The vsrc captures in sync with the playback in sink.

As in example 3, sink can only be set to PLAYING after it successfully prerolled.

State Changes

A sink is never set to PLAYING before it is prerolled. In order to do this, the pipeline (at the GstBin level) keeps track of all elements that require preroll (the ones that return ASYNC from the state change). These elements posted an ASYNC_START message without a matching ASYNC_DONE one.

The pipeline will not change the state of the elements that are still doing an ASYNC state change.

When an ASYNC element prerolls, it commits its state to PAUSED and posts an ASYNC_DONE message. The pipeline notices this ASYNC_DONE message and matches it with the ASYNC_START message it cached for the corresponding element.

When all ASYNC_START messages are matched with an ASYNC_DONE message, the pipeline proceeds with setting the elements to the final state again.

The base time of the element was already set by the pipeline when it changed the NO_PREROLL element to PLAYING. This operation has to be performed in the separate async state change thread (like the one currently used for going from PAUSED→PLAYING in a non-live pipeline).


The pipeline latency is queried with the LATENCY query.

  • live G_TYPE_BOOLEAN (default FALSE): - if a live element is found upstream

  • min-latency G_TYPE_UINT64 (default 0, must not be NONE): - the minimum latency in the pipeline, meaning the minimum time downstream elements synchronizing to the clock have to wait until they can be sure all data for the current running time has been received.

Elements answering the latency query and introducing latency must set this to the maximum time for which they will delay data, while considering upstream's minimum latency. As such, from an element's perspective this is not its own minimum latency but its own maximum latency. Considering upstream's minimum latency generally means that the element's own value is added to upstream's value, as this will give the overall minimum latency of all elements from the source to the current element:

min_latency = upstream_min_latency + own_min_latency
  • max-latency G_TYPE_UINT64 (default 0, NONE meaning infinity): - the maximum latency in the pipeline, meaning the maximum time an element synchronizing to the clock is allowed to wait for receiving all data for the current running time. Waiting for a longer time will result in data loss, buffer overruns and underruns and, in general, breaks synchronized data flow in the pipeline.

Elements answering the latency query should set this to the maximum time for which they can buffer upstream data without blocking or dropping further data. For an element, this value will generally be its own minimum latency, but might be bigger than that if it can buffer more data. As such, queue elements can be used to increase the maximum latency.

The value set in the query should again consider upstream's maximum latency:

  • If the current element has blocking buffering, i.e. it does not drop data by itself when its internal buffer is full, it should just add its own maximum latency (i.e. the size of its internal buffer) to upstream's value. If upstream's maximum latency, or the elements internal maximum latency was NONE (i.e. infinity), it will be set to infinity.
if (upstream_max_latency == NONE || own_max_latency == NONE)
  max_latency = NONE;
  max_latency = upstream_max_latency + own_max_latency;

If the element has multiple sinkpads, the minimum upstream latency is the maximum of all live upstream minimum latencies.

If the current element has leaky buffering, i.e. it drops data by itself when its internal buffer is full, it should take the minimum of its own maximum latency and upstream’s. Examples for such elements are audio sinks and sources with an internal ringbuffer, leaky queues and in general live sources with a limited amount of internal buffers that can be used.

max_latency = MIN (upstream_max_latency, own_max_latency)

Note: many GStreamer base classes allow subclasses to set a minimum and maximum latency and handle the query themselves. These base classes assume non-leaky (i.e. blocking) buffering for the maximum latency. The base class' default query handler needs to be overridden to correctly handle leaky buffering.

If the element has multiple sinkpads, the maximum upstream latency is the minimum of all live upstream maximum latencies.


The latency in the pipeline is configured with the LATENCY event, which contains the following fields:

  • latency G_TYPE_UINT64: the configured latency in the pipeline

Latency compensation

Latency calculation and compensation is performed before the pipeline proceeds to the PLAYING state.

When the pipeline collected all ASYNC_DONE messages it can calculate the global latency as follows:

  • perform a latency query on all sinks
  • sources set their minimum and maximum latency
  • other elements add their own values as described above
  • latency = MAX (all min latencies)
  • if MIN (all max latencies) < latency, we have an impossible situation and we must generate an error indicating that this pipeline cannot be played. This usually means that there is not enough buffering in some chain of the pipeline. A queue can be added to those chains.

The sinks gather this information with a LATENCY query upstream. Intermediate elements pass the query upstream and add the amount of latency they add to the result.

ex1: sink1: [20 - 20] sink2: [33 - 40]

    MAX (20, 33) = 33
    MIN (20, 40) = 20 < 33 -> impossible

ex2: sink1: [20 - 50] sink2: [33 - 40]

    MAX (20, 33) = 33
    MIN (50, 40) = 40 >= 33 -> latency = 33

The latency is set on the pipeline by sending a LATENCY event to the sinks in the pipeline. This event configures the total latency on the sinks. The sink forwards this LATENCY event upstream so that intermediate elements can configure themselves as well.

After this step, the pipeline continues setting the pending state on its elements.

A sink adds the latency value, received in the LATENCY event, to the times used for synchronizing against the clock. This will effectively delay the rendering of the buffer with the required latency. Since this delay is the same for all sinks, all sinks will render data relatively synchronised.

Flushing a playing pipeline

We can implement resynchronisation after an uncontrolled FLUSH in (part of) a pipeline in the same way. Indeed, when a flush is performed on a PLAYING live element, a new base time must be distributed to this element.

A flush in a pipeline can happen in the following cases:

  • flushing seek in the pipeline

  • performed by the application on the pipeline

  • performed by the application on an element

  • flush preformed by an element

  • after receiving a navigation event (DVD, …)

When a playing sink is flushed by a FLUSH_START event, an ASYNC_START message is posted by the element. As part of the message, the fact that the element got flushed is included. The element also goes to a pending PAUSED state and has to be set to the PLAYING state again later.

The ASYNC_START message is kept by the parent bin. When the element prerolls, it posts an ASYNC_DONE message.

When all ASYNC_START messages are matched with an ASYNC_DONE message, the bin will capture a new base_time from the clock and will bring all the sinks back to PLAYING after setting the new base time on them. It’s also possible to perform additional latency calculations and adjustments before doing this.

Dynamically adjusting latency

An element that wants to change the latency in the pipeline can do this by posting a LATENCY message on the bus. This message instructs the pipeline to:

  • query the latency in the pipeline (which might now have changed) with a LATENCY query.

  • redistribute a new global latency to all elements with a LATENCY event.

A use case where the latency in a pipeline can change could be a network element that observes an increased inter-packet arrival jitter or excessive packet loss and decides to increase its internal buffering (and thus the latency). The element must post a LATENCY message and perform the additional latency adjustments when it receives the LATENCY event from the downstream peer element.

In a similar way, the latency can be decreased when network conditions improve.

Latency adjustments will introduce playback glitches in the sinks and must only be performed in special conditions.

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